[Asterisk-Users] Where is the difference sip.conf - Real-time ?

[Ronald Wiplinger]

Alban wrote:
> Hello,
> Have you tried to delete rtnoupdate and rtautoclear lines? I'm using realtime, 
> with caching for sip but without those 2 lines, and works perfectly.
> Another point : verify that you have the field fullcontact in your realtime 
> sip table.
> Bye,
> Alban Elziere
>   

While I compiled the message, I discovered the difference already in 
canreinvite=yes/no
I could test it, and it was the problem!
I found than that if you have the phones behind asterisk you MUST have 
canreinvite=no to force, the rtp stream to go through asterisk and not 
to try to bypass it. I use bypass so that the users are directly 
connected to the gateways without bothering my servers bandwidth.

rtnoupdate=yes
 ; do not send the update request over realtime.

rtautoclear=yes
 ; Auto-Expire friends created on the fly on the same schedule
 ; as if it had just registered when the registration expires
 ; the friend will vanish from the configuration until requested
 ; again.  If set to an integer, friends expire
 ; within this number of seconds instead of the
 ; same as the registration interval


>> I have two phones (111 and 112) on a LAN, and I have on a users site a
>> phone 333.
>>
>> phone 111 uses sip.conf, while 112 uses real-time set-up.
>> 111 can call 333 AND the audio is working
>> 112 can call 333 but audio is just white noise.
>> 333 can call 111 or 112 and audio is working.
>> The phones are identically set-up (just user name = phone number and
>> password are different)
>>
>> sip.conf (for 111 - all remarked lines removed)
>>
>> [general]
>> context=default            ; Default context for incoming calls
>> port=5060            ; UDP Port to bind to (SIP standard port is 5060)
>> bindaddr=0.0.0.0        ; IP address to bind to (0.0.0.0 binds to all)
>> srvlookup=yes            ; Enable DNS SRV lookups on outbound calls
>> tos=lowdelay                   ;
>> lowdelay,throughput,reliability,mincost,none
>> maxexpirey=7200        ; Max length of incoming registration we allow
>> defaultexpirey=3600        ; Default length of incoming/outoing
>> registration videosupport=yes        ; Turn on support for SIP video
>> disallow=all            ; First disallow all codecs
>> allow=ulaw            ; Allow codecs in order of preference
>> allow=alaw
>> allow=g729
>> allow=gsm
>> rtcachefriends=yes
>> rtnoupdate=yes
>>  rtautoclear=yes
>> externip = 59.14.2.1
>> localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
>>
>>
>> [111]
>> type=friend
>> username=hotline
>> secret=I-know-it
>> canreinvite=no
>> host=dynamic
>> dtmfmode=rfc2833
>> mailbox=111 at default
>> nat=yes
>> callgroup=1
>> pickupgroup=1
>> callerid="Ronald Hotline",<111>
>> qualify=1000
>>
>>
>> Real-time for 112:
>> name=112
>> callerid="Ronald Hotline",<112>
>> canreinvite=yes
>> context=default
>> dtmfmode=rfc2833
>> host=dynamic
>> language=en
>> mailbox=112 at default
>> nat=yes
>> pickupgroup=1
>> port=5060
>> qualify=1000
>> secret=I-know-it
>> type=friend
>> username=112
>> disallow=all
>> allow=ulaw;alaw;g729;gsm
>> cancallforward=yes
>>
>>
>> Which of the settings cause the different behaviour?
>> Which settings should I change (maybe not related to the problem)?
>>
>> bye
>>
>> Ronald
>>
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>   


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